Trim AIF — Free Online Tool
Trim an AIF audio file to an exact time range while preserving the original lossless PCM audio stream — no re-encoding, no quality loss. Ideal for cutting high-quality uncompressed audio used in professional Mac-based workflows without altering the pcm_s16be (or higher bit-depth) data.
to
FFmpeg Command
Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg
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Note: Browser-based encoding uses approximate quality targets. For precise CRF compression, copy the FFmpeg command above and run it on your desktop.
Estimated output:
Conversion Complete!
DownloadHow It Works
Because both the input and output are AIF containers sharing the same PCM codec family (pcm_s16be, pcm_s24be, etc.), FFmpeg performs a stream copy rather than a re-encode. The audio data between your specified start and end timestamps is extracted and written directly into a new AIF container. Since PCM audio has no keyframe dependency — every sample is independently decodable — stream copying works cleanly at any cut point without the frame-boundary complications that affect compressed formats like AAC or MP3. The result is a bit-for-bit accurate segment of your original uncompressed audio.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg command-line tool. When run in the browser, this is executed via FFmpeg.wasm, a WebAssembly port that processes your AIF file entirely locally without any server upload. |
-i input.aif
|
Specifies the input AIF file containing the uncompressed PCM audio to be trimmed. FFmpeg reads the container structure and identifies the PCM audio stream (e.g., pcm_s16be or pcm_s24be) before applying the trim parameters. |
-ss 00:00:00
|
Sets the trim start point at the beginning of the file (0 seconds). Placed after '-i', this performs a fast seek followed by precise sample-level trimming — safe for PCM audio since every sample is independently decodable. |
-to 00:00:10
|
Sets the trim end point at 10 seconds into the AIF file. Only the audio between the '-ss' start time and this timestamp will be included in the output. For PCM audio, this is sample-accurate at any timestamp. |
-c copy
|
Instructs FFmpeg to copy the PCM audio stream directly into the output AIF container without re-encoding. This preserves the exact bit depth, sample rate, and audio data of the original, and makes the operation nearly instantaneous since no decoding or encoding is performed. |
output.aif
|
Defines the output filename and container format. The .aif extension tells FFmpeg to write the trimmed PCM stream into an AIF container, maintaining full compatibility with macOS applications, Logic Pro, Pro Tools, and other professional audio tools that expect the AIF format. |
Common Use Cases
- Isolate a specific musical phrase or take from a full uncompressed recording session for delivery to a mixing engineer who requires lossless AIF files
- Trim a long field recording captured to AIF on a Mac to remove silence or unwanted content before importing into Logic Pro or Pro Tools
- Extract a precise clip from an AIF sound effect library file to create a shorter asset sized for a specific game engine sample slot
- Cut a defined segment from an AIF master file for A/B comparison testing without touching the original archive
- Remove a flawed section at the beginning or end of a vocal recording stored as AIF before handing off to a podcast editor
- Trim an AIF reference track to match a specific song or scene length for synchronization work in a video production timeline
Frequently Asked Questions
No. Because AIF uses uncompressed PCM audio, FFmpeg can copy the audio stream directly without decoding and re-encoding it. The trimmed output contains the exact same PCM sample values as the corresponding portion of the original file — there is zero generational quality loss regardless of whether the source is 16-bit, 24-bit, or 32-bit.
Yes. The '-c copy' flag instructs FFmpeg to pass the audio stream through unchanged, preserving the original codec (e.g., pcm_s24be for 24-bit big-endian PCM), sample rate, and channel layout. No transcoding or resampling occurs during the trim operation.
Compressed audio formats like MP3 store data in frames where each frame depends on context from surrounding frames, so cuts must align to frame boundaries or cause audible artifacts. PCM audio in AIF has no such dependency — every individual sample is fully independent. This means a stream-copy trim in AIF is sample-accurate at any arbitrary timestamp, with no need to re-encode to achieve a clean edit point.
Modify the values after '-ss' (start time) and '-to' (end time) using the format HH:MM:SS or HH:MM:SS.mmm for millisecond precision. For example, to trim from 30 seconds to 1 minute and 15.5 seconds, use '-ss 00:00:30 -to 00:01:15.500'. You can also use '-t' instead of '-to' to specify a duration rather than an absolute end time — for instance, '-t 00:00:45' would extract 45 seconds starting from the '-ss' point.
Yes. On macOS or Linux you can use a shell loop: 'for f in *.aif; do ffmpeg -i "$f" -ss 00:00:00 -to 00:00:10 -c copy "trimmed_$f"; done'. On Windows Command Prompt use: 'for %f in (*.aif) do ffmpeg -i "%f" -ss 00:00:00 -to 00:00:10 -c copy "trimmed_%f"'. Since stream copying is used, each file processes very quickly regardless of its original size.
AIF has limited metadata support compared to formats like FLAC or MP4. FFmpeg will attempt to copy any embedded metadata chunks present in the original AIF container, but AIF's metadata capabilities are basic and some DAW-specific chunks may not be retained. If precise metadata preservation is critical, verify the output in your target application after trimming.
Technical Notes
AIF (Audio Interchange File Format) stores audio as big-endian PCM data, which is why its codec identifiers carry the 'be' suffix — pcm_s16be (16-bit signed big-endian), pcm_s24be (24-bit), pcm_s32be (32-bit integer), pcm_f32be (32-bit float), and pcm_f64be (64-bit float). Because the tool uses '-c copy', the output AIF will use whichever of these codecs is present in the source file — no downsampling or conversion occurs. One practical consideration: AIF files are uncompressed, so a 10-second trim of a 24-bit/96kHz stereo file will produce an output proportionally smaller than the source but still substantially larger than an equivalent lossy format. AIF does not support subtitles, chapters, or multiple audio tracks, so no such data can be lost during trimming. The format is natively supported by macOS, Logic Pro, Pro Tools, GarageBand, and most professional DAWs, making the trimmed output immediately usable in those environments without any compatibility concerns.